Is it legitimate to call my Bible study software as MacBible?

I am a developer of a Bible study software. Can I call it MacBible and not be sued from Apple?

Ok but no one here can answer a legal question. This is a user to user forum, not Apple.
Pardon the pun but you need to talk to a higher power.
Apple, Inc.
1 Infiniti Loop
Cupertino, CA  
95014
It appears you may have some competition >  http://www.macbiblesoftware.com/

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    I think I have a working solution for a system-wide install with shared books.
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  • Bible study software done right: ISA2

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    17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_TRYING value=500 retries=3|2,100,230,1.4901096^ISP-IP^*
    17:34:18.536 |//SIP/Stack/States/0xb1b50c90/0xb1b50c90 : State change from (STATE_SENT_INVITE, SUBSTATE_NONE)  to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)|2,100,230,1.4901096^ISP-IP^*
    17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_EXPIRES value=180000 retries=0|2,100,230,1.4901096^ISP-IP^*
    17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_start_timer: type=SIP_TIMER_EXPIRES value=180000 retries=0|2,100,230,1.4901096^ISP-IP^*
    17:34:18.561 |EnvProcessUdpHandler::handle_input - handle = 334|*^*^*
    17:34:18.561 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 2|*^*^*
    17:34:18.561 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 396 from ISP-IP:[5060]:
    [12623362,NET]
    SIP/2.0 403 Forbidden
    Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
    CSeq: 101 INVITE
    From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
    To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
    Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
    Server: CISCO-SBC/2.x
    Content-Length: 0
    Contact: <sip:ISP-IP:5060>
    [12623363,NET]
    ACK sip:2484XXX@ISP-IP:5060 SIP/2.0
    Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
    From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
    To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
    Date: Wed, 18 Dec 2013 13:34:18 GMT
    Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: presence
    Content-Length: 0
    INVITE sip:2484XXX@ISP's-Other-IP:5062 SIP/2.0
    Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900
    From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
    To: <sip:2484XXX@ISP's-Other-IP>
    Date: Wed, 18 Dec 2013 13:34:18 GMT
    Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM8.6
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 101 INVITE
    Expires: 180
    Allow-Events: presence
    Supported: X-cisco-srtp-fallback
    Supported: Geolocation
    Cisco-Guid: 0383266432-0000065536-0000191816-2219117834
    Session-Expires:  1800
    P-Asserted-Identity: <sip:057729XXXX@MY-CUCM-IP>
    Remote-Party-ID: <sip:057729XXXX@MY-CUCM-IP>;party=calling;screen=yes;privacy=off
    Contact: <sip:057729XXXX@MY-CUCM-IP:5062>
    Max-Forwards: 68
    Content-Type: application/sdp
    Content-Length: 215
    v=0
    o=CiscoSystemsCCM-SIP 4052092 1 IN IP4 MY-CUCM-IP
    s=SIP Call
    c=IN IP4 MY-CUCM-IP
    t=0 0
    m=audio 29792 RTP/AVP 8 101
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    |2,100,56,1.173711431^MY-CUCM-IP^MTP_3
    17:34:18.567 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 0|2,100,56,1.173711431^MY-CUCM-IP^MTP_3
    17:34:18.567 |EnvProcessUdpHandler::fireSignal - SEND: index = 0, handler = 0xa6b4d7c0|*^*^*
    17:34:18.567 |EnvProcessUdpPort::fireSignal - SEND, destination = ISP's-Other-IP:5062|*^*^*
    17:34:18.567 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 1177, ISP's-Other-IP:5062)|*^*^*
    17:34:18.569 |EnvProcessUdpHandler::handle_input - handle = 335|*^*^*
    17:34:18.569 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 0|*^*^*
    17:34:18.569 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 394 from ISP's-Other-IP:[5062]:
    [12623365,NET]
    SIP/2.0 100 trying -- your call is important to us
    Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900;rport=5062
    From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
    To: <sip:2484XXX@ISP's-Other-IP>
    Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
    CSeq: 101 INVITE
    Server: kamailio (3.3.1 (x86_64/linux))
    Content-Length: 0
    17:34:18.587 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 375 from ISP's-Other-IP:[5062]:
    [12623366,NET]
    SIP/2.0 403 Forbidden
    Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900;rport=5062
    Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
    From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
    To: <sip:2484XXX@ISP's-Other-IP>;tag=dc6a4ae7
    CSeq: 101 INVITE
    Reason: Q.850;cause=0;text="unknown"
    Content-Length: 0
    |2,100,230,1.4901099^ISP's-Other-IP^*
    [12623367,NET]
    ACK sip:2484XXX@ISP's-Other-IP:5062 SIP/2.0
    Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900
    From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
    To: <sip:2484XXX@ISP's-Other-IP>;tag=dc6a4ae7
    Date: Wed, 18 Dec 2013 13:34:18 GMT
    Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: presence
    Content-Length: 0

    SIP/2.0 403 Forbidden error
    If your router is sending a SIP/2.0 403 Forbidden error to the SIP server you are registered to, there is a good chance your  router is blocking the incoming call due to the toll-faud prevention  feature that was added to IOS version 15.1(2)T.
    How to Identify if TOLLFRAUD_APP is Blocking Your Call
    If the TOLLFRAUD_APP is rejecting the call, it generates a Q.850       disconnect cause value of 21, which represents ‘Call Rejected’. The       debug voip ccapi inout command can be run to       identify the cause value.
    Additionally, voice iec syslog can be       enabled to further verify if the call failure is a result of the toll-fraud       prevention. This configuration, which is often handy to troubleshoot the origin       of failure from a gateway perspective, will print out that the call is being       rejected due to toll call fraud. The CCAPI and Voice IEC output is demonstrated       in this debug output:
    %VOICE_IEC-3-GW: Application Framework Core: Internal Error (Toll fraud call rejected):
    IEC=1.1.228.3.31.0 on callID 3 GUID=F146D6B0539C11DF800CA596C4C2D7EF
    000183: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/ccCallSetContext:
       Context=0x49EC9978
    000184: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/cc_process_call_setup_ind:
       >>>>CCAPI handed cid 3 with tag 1002 to app "_ManagedAppProcess_TOLLFRAUD_APP"
    000185: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/ccCallDisconnect:
       Cause Value=21, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
    The Q.850 disconnect value that is returned for blocked calls can also       be changed from the default of 21 with this command:
    voice service voip
    ip address trusted call-block cause
    How to Return to Pre-15.1(2)T Behavior
    Source IP Address Trust List
    There are three ways to return to the previous behavior of voice       gateways before this trusted address toll-fraud prevention feature was       implemented. All of these configurations require that you are already running       15.1(2)T in order for you to make the configuration change.
    Explicitly enable those source IP addresses from which you would like           to add to the trusted list for legitimate VoIP calls. Up to 100 entries can be           defined. This below configuration accepts calls from those host           203.0.113.100/32, as well as from the network 192.0.2.0/24. Call setups from           all other hosts are rejected. This is the recommended method from a voice           security perspective.
    voice service voip
    ip address trusted list
      ipv4 203.0.113.100 255.255.255.255
      ipv4 192.0.2.0 255.255.255.0
    Configure the router to accept incoming call setups from all source           IP addresses.
    voice service voip
    ip address trusted list
      ipv4 0.0.0.0 0.0.0.0
    Disable the toll-fraud prevention application completely.
    voice service voip
    no ip address trusted authenticate
    Two-Stage Dialing
    If two-stage dialing is required, the following can be configured to       return behavior to match previous releases.
    For inbound ISDN calls:
    voice service pots
    no direct-inward-dial isdn
    For inbound FXO calls:
    voice-port
    secondary dialtone

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